[asterisk-users] *8 and SIP

Tue Dec 31 13:22:38 CST 2013

On 12/31/13, 11:23 AM, Nick Olsen wrote:
> Greetings all, First time poster, Sorry if this has been answered here 
> before.
>
> We recently replaced a failed 1.4x asterisk PBX at a customer location.
>
> Voicemail access was setup when the customer dialed *8, This worked in 
> 1.4.
I suggest trying command 'features show' to pinpoint the conflict.

# asterisk -rx 'features show'

Builtin Feature           Default Current
---------------           ------- -------
Pickup                    *8
Blind Transfer            #       #
Attended Transfer
One Touch Monitor
Disconnect Call           *       *
Park Call
One Touch MixMonitor

Dynamic Feature           Default Current
---------------           ------- -------
(none)

Feature Groups:
---------------
(none)

Call parking (Parking lot: default)
------------
Parking extension     :      700
Parking context       :      parkedcalls
Parked call extensions:      701-750
Parkingtime           :      45000 ms
MusicOnHold class     :      default
Enabled               :      Yes

>
> Now, Running 1.6 (I know it's old I had to load it quickly, And that's 
> what I got working first. It'll get upgraded to 1.8 soon).
>
> The strange part is *8 no longer works.
> The only CLI feedback I get is "== Using SIP RTP CoS mark 5"
>
> In features.conf, Callpickup *8 is commented out, But just incase I 
> also changed it to *7 (We don't use that feature).
>
> It appears to be something completely SIP based, As if the call 
> originates from DAHDI, It works fine..
>
> If anyone has any ideas, Please let me know. Thanks!
>
> SIP Trace Below
>
> <--- SIP read from UDP:208.65.55.170:5063 --->
> INVITE sip:*8 at 10.65.6.10:5060 SIP/2.0
> Via: SIP/2.0/UDP 172.16.10.101:5063;branch=z9hG4bK908225576
> From: "nicktest" <sip:nicktest at 10.65.6.10>;tag=1470823868
> To: <sip:*8 at 10.65.6.10>
> Call-ID: 695101044 at 172.16.10.101
> CSeq: 1 INVITE
> Contact: <sip:nicktest at 172.16.10.101:5063>
> Content-Type: application/sdp
> Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, 
> REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
> Max-Forwards: 70
> User-Agent: Yealink SIP-T46G 28.71.0.180
> Supported: replaces
> Allow-Events: talk,hold,conference,refer,check-sync
> Content-Length: 308
>
> v=0
> o=- 20402 20402 IN IP4 172.16.10.101
> s=SDP data
> c=IN IP4 172.16.10.101
> t=0 0
> m=audio 11792 RTP/AVP 0 8 18 9 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:9 G722/8000
> a=fmtp:101 0-15
> a=rtpmap:101 telephone-event/8000
> a=ptime:20
> a=sendrecv
>
> <------------->
> --- (14 headers 15 lines) ---
>   == Using SIP RTP CoS mark 5
> Using INVITE request as basis request - 695101044 at 172.16.10.101
> Found peer 'nicktest' for 'nicktest' from 208.65.55.170:5063
> Found RTP audio format 0
> Found RTP audio format 8
> Found RTP audio format 18
> Found RTP audio format 9
> Found RTP audio format 101
> Found audio description format PCMU for ID 0
> Found audio description format PCMA for ID 8
> Found audio description format G729 for ID 18
> Found audio description format G722 for ID 9
> Found audio description format telephone-event for ID 101
> Capabilities: us - 0x4 (ulaw), peer - audio=0x110c 
> (ulaw|alaw|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined 
> - 0x4 (ulaw)
> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 
> (telephone-event), combined - 0x1 (telephone-event)
> Peer audio RTP is at port 172.16.10.101:11792
> Looking for *8 in trunk_office (domain 10.65.6.10)
> list_route: hop: <sip:nicktest at 172.16.10.101:5063>
>
> <--- Transmitting (NAT) to 208.65.55.170:5063 --->
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 
> 172.16.10.101:5063;branch=z9hG4bK908225576;received=208.65.55.170
> From: "nicktest" <sip:nicktest at 10.65.6.10>;tag=1470823868
> To: <sip:*8 at 10.65.6.10>
> Call-ID: 695101044 at 172.16.10.101
> CSeq: 1 INVITE
> Server: Asterisk PBX 1.6.2.20
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> Supported: replaces, timer
> Contact: <sip:*8 at 10.65.6.10>
> Content-Length: 0
>
>
> <------------>
> Scheduling destruction of SIP dialog '695101044 at 172.16.10.101' in 6400 
> ms (Method: INVITE)
>
> <--- Reliably Transmitting (NAT) to 208.65.55.170:5063 --->
> SIP/2.0 403 Forbidden
> Via: SIP/2.0/UDP 
> 172.16.10.101:5063;branch=z9hG4bK908225576;received=208.65.55.170
> From: "nicktest" <sip:nicktest at 10.65.6.10>;tag=1470823868
> To: <sip:*8 at 10.65.6.10>;tag=as65ceb9be
> Call-ID: 695101044 at 172.16.10.101
> CSeq: 1 INVITE
> Server: Asterisk PBX 1.6.2.20
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> Supported: replaces, timer
> Content-Length: 0
>
>
> <------------>
>
> <--- SIP read from UDP:208.65.55.170:5063 --->
> ACK sip:*8 at 10.65.6.10:5060 SIP/2.0
> Via: SIP/2.0/UDP 172.16.10.101:5063;branch=z9hG4bK908225576
> From: "nicktest" <sip:nicktest at 10.65.6.10>;tag=1470823868
> To: <sip:*8 at 10.65.6.10>;tag=as65ceb9be
> Call-ID: 695101044 at 172.16.10.101
> CSeq: 1 ACK
> Content-Length: 0
>
>
> <------------->
>
> Nick Olsen
> Network Operations
> (855) FLSPEED  x106
>
>
>


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